Fiter design is the process of designing a filter.

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Low-pass elliptic filter design: equation for calculating required minimum order

In this paper on page 4 (or in slide page, 23), it describes the equation for calculating the minimum required order for elliptic filter for design specification. However, it says it is only the ...
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0answers
15 views

Anti tremor motion detection

I'm programming in c# using opencv (emguCV). I use motion detection approach to detect some changes in a paper wall. For example it's may be colored by a laser pointer. My problem is that when I use ...
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0answers
32 views

Designing bandpass FIR filter MATLAB

I am trying to design a FIR bandpass filter for a STM32F407 microcontroller with a passband of 1kHz between 59500 Hz and 60500 Hz or something similar (since I don't really know how tight I can ...
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4answers
108 views

Why is a linear phase important?

If symmetry conditions are met, FIR filters have a linear phase. This is not true for IIR filters. However, for what applications is it bad to apply filters that do not have this property and what ...
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1answer
29 views

How to construct a preemptive digital filter to neutralize an unwanted filter

I'm trying to think of how to construct a filter based on the following scenario. I have some time trace with a certain known power spectral density (I can verify this with a periodogram of the time ...
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39 views

Analytically Deriving butterworth Filter [closed]

I am pretty new to Digital Signal processing, I did my back end research, but I was overwhelmed by the variety of the solutions I got. I want to Analytically derive a Butterworth filter with only "...
2
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1answer
63 views

Decomposition of $H(z)$ as maximum-phase, minimum-phase

The frequency response is: $$H(z) = 2-7z^{-1}+7z^{-2}-2z^{-3}$$ I see that it has $3$ zeros: $z_{01} = \frac 12$, $z_{02} = 2$, and $z_{03} = 1$; and $3$ poles in: $$z_x = 0$$ Now, I have to write ...
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1answer
38 views

Maximum phase with the same magnitude response

I'm trying to solve an exercise in a general way and I can't find if my answer is correct. Given this frequency response: $$H(z) = 4 + 2\sqrt2 z^{-1} + z^{-2}$$ I need to find a frequency response ...
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1answer
45 views

Testing filter code w/ Octave

I want to test some code generated by this site. I selected Bessel LP 1st sample rate $600\textrm{ Hz}$ corner $8\textrm{ Hz}$ long $10\textrm{ bit}$. If I adjust the code for octave to be: ...
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1answer
86 views

Weighting function for output error IIR filter design

I noticed, when I try to fit filter coefficients to a given complex transfer function with the output error method, implemented for example in the MATLAB function ...
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1answer
26 views

What is an 'Oriented' Band-pass Filter?

In this research paper, Section 4.1 talks about a filter called "Oriented Band-pass Filter". What does it actually mean by 'Orientation'? Also, the article gives a function, $$H(u, v) = \frac{1}{1 +...
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1answer
55 views

Why can adding delay improve the phase fit in fitting complex transfer functions?

When you fit an IIR filter to a complex transfer function you can use a delay to get a stable filter and improve the fitting results in your phase response. Can anyone explain me the reason for this ...
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1answer
35 views

Extracting filter coefficients from frequency response

The frequency response of filter transfer function is given as $$H(j\omega)= \begin{cases} 0 &, \textrm{if }\quad (1+r)\frac{\pi}{2} < \lvert \omega \rvert < \pi \\ 1 + \cos \left ( \...
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1answer
26 views

How many types of Bandpass filters are there for image processing?

This article shows that: Ideal Bandpass filter Butterworth Bandpass filter Gaussian Bandpass filter Is that classification correct? Are there any other types?
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1answer
32 views

What is Local Mean Filter?

The research paper "Multidirectional Scratch Detection and Restoration in Digitized Old Images" says that, 4.1. Preprocessing. The preprocessing step aims to enhance image features along a set of ...
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1answer
44 views

What is a filter bank?

I have three Gabor filters. I am applying them to an image serially. Is this the correct concept of a Filter Bank? What if I want to apply the filter in parallel? I am using Accord.NET framework. ...
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0answers
48 views

Upsampling with FIR filter causing discontinuity at start of new buffer [closed]

I have implemented an oversampling algorithm for an audio signal in C++ which is producing a distortion of the signal at the start of a buffer. The algorithm is wrapped in a class with methods for up ...
3
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1answer
63 views

Basic tools for digital filter design?

What tools to use for practicing elementary filter design? Is MATLAB all there is? Do I need some specific toolboxes? What functions do I need? I'm starting from the ground up in digital filter ...
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5answers
201 views

How many taps does an FIR filter need?

I am looking to design a set of FIR filters to implement a low pass filter. I am also trying to reduce the latency of the signal through the filter so I am wondering what the minimum number of taps I ...
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0answers
18 views

What is the multi-resolution analysis form of Gabor filter?

The multi-resolution analysis form of Gabor filter is as follows: \begin{align} \psi\left(x,y;f_0,\theta\right)&=\frac{f_0^2}{\pi\gamma\eta}\exp\left[-\left(\frac{f_0^2}{\gamma^2}x'^2+\frac{f_0^2}...
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1answer
32 views

When should we use the full Gabor Function and when do we use only the real part?

In this YouTube video, we see that the guy is only using the real part of the Gabor Filter to implement his project. My questions are, When should we use the complex Gabor Function? When should we ...
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0answers
25 views

Deriving the multi-resolution analysis form of Gabor filter from the standard form

The standard form of a 2D Gabor filter is as follows: \begin{align} g_{\lambda, \theta,\varphi, \sigma,\gamma}(x, y)&=\exp\left(-\frac{x^2+\gamma^2y^2}{2\sigma^2}\right)\cos\left(2\pi\frac{x'}{\...
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2answers
48 views

Goodness of fit for complex valued curves (i.e. frequency responses in frequency domain)

My apologies for perhaps the stupidity of this question. Presume that one has the 'frequency response' $Y_{data}(k)$ of a system and also has an estimated model $Y_{syn}(k)$ that fits the data. How ...
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0answers
23 views

anti aliasing filter which suppresses spectral components above half sampling frequency

I'm new to designing filters, and I have to make an anti aliassing filter (in python) in function of the subsampling factor, which suppresses spectral components above half the sampling frequency with ...
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2answers
57 views

Lowpass Butterworth filter design (own poles' calculations)

I'm trying to calculate coefficients for my Butterworth filter of my own. I've found this article that looks really reliable. But I'm trying to implement it in MATLAB and I'm getting bandstop filter. ...
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1answer
43 views

Hilbert FIR Filter for Q - Matching Filter for I

For an ultrasound application, I am collecting data from an ADC using high spec Altera FPGA which grabs data and can then do some DSP processing that would be otherwise too slow for a desktop to do. ...
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1answer
28 views

How are these pole zero plots created

Back around 5 BC, where C stands for the programming language, engineers would photograph their oscilloscope's display to record their test results. This appears to be such a photograph, taken from ...
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0answers
26 views

Stopband Attenuation

Why is a stopband attenuation of -40dB preferred for the filter design & analysis. I understand at this magnitude response point, the o/p is just 0.01 of the O/P but how does it help in defining ...
3
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1answer
31 views

Can we determine if a filter is butterworth or chebyshev from its physical topology?

Can all analog filters be classified as Bessel, Elliptic, Butterworth or Chebyshev? Given a physical ladder topology of several stages of {L, C or LC} in {series or parallel}, is it possible to ...
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1answer
56 views

How can a signal have two maximum frequency components?

$x(t)$ can be exactly reconstructed from its samples at $\omega_s = 10 \textrm{ rad/sec}$. My conclusion is that the maximum frequency component in $x(t)$ is $5\textrm{ rad/sec}$. But I'm being told ...
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2answers
60 views

Analytical way to generate an IIR that is the square root response of another IIR

For simplicity take the transfer function of a second order IIR in $z$-domain: $$ H(z) = \frac {b_0 + b_1 z^{-1} + b_2 z^{-2}} {a_0 + a_1 z^{-1} + a_2 z^{-2}} $$ Is it possible to generate a new ...
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1answer
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describing a block diagram // How do I describe multiplying a signal through multiple branches formally?

Let's say I made this block diagram and I want to explain it: FYI: $x$ is a signal and each $y$ box is a matrix I want to say that: The signal $x$ is multiplied by each matrix $y$ in the ...
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2answers
70 views

What is the difference between a'mixer' and a ' multiplier' used in modulation process?

I am not sure if it is valid to conclude that a mixer produces a difference/sum of frequencies at output ,while multiplier produces multiples of frequency at its output. I am keen to know how far I ...
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0answers
16 views

How to design Bi orthogonal wavelet filter bank

I know in orthogonal wavelet system analysis filters can be made by flipping the synthesis filters(vice versa) but in bi orthogonal system this will not work because the co efficient are symmetric in ...
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33 views

FIR anti aliasing filter design

Suppose we have a signal with a $3\mathrm{kHz}$ bandwidth and a sampling rate of $48\mathrm{kHz}$. Then we want to decimate ($6$-rate decimator) it, in which the decimator has a passband ripple of $1\...
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1answer
41 views

Filter Design near Nyquist frequency (complex curve fitting)

I'm fitting IIR filters to a complex transfer function with Matlab's invfreqz.m. My fitting results are very good, but when I come very close to the Nyquist frequency the phase response runs to 0 and ...
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2answers
36 views

Wavelet filter coefficients from scaling filter coefficients

I am trying to develop a new type of wavelets and I found out a function that following a particular two scale relation. The function at a scale $t$ say $x(t)$ can be related in a finer scale $x(2t)$ ...
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88 views

How to design a variable slope highpass filter

I want to design some specific slope highpass filter to enhance speech; just like $1$ dB/octave, $2$ dB/octave and more. Is there a method to calculate the coefficients by specific slope?
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47 views

Pre-emphasis filter design

I am using the standard MATLAB filter function to design pre-emphasis filter. However, I am getting the wrong results back! I have the ground truth so I can check ...
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3answers
87 views

Continuous phase for phase delay calculus in FIR filters

I would like to estimate the phase delay accurately for any random FIR filter. The definition of the phase delay is the continuous phase divided by the angular frequency (with a sign change). That ...
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1answer
56 views

Very-short Hilbert transform FIR (or “recursive”) approximations

Hilbert transform is a quite sensitive topic here, since Gabor's paper, Theory of Communication, J. Inst. Electr. Engineering, London, 1946. Perhaps even more important than the Fourier transform. ...
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45 views

Combining IIM and MZT transformation

Split from thread: Correcting Impulse Invariance Method Back to this subject. Matt L suggested combining IIM and MZT b-coeffs to get the filter response improved at hihger frequency area. Couldn't ...
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19 views

Generating 3-dimension $1/f^\alpha$ noise

I'm stumped trying to find a way to generate a 3-dimensional $1/f^\alpha$ noise time series. Basically I want simulate a forcing that is $1/f^\alpha$ in the 2 space dimensions and $1/f^\beta$ in time, ...
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1answer
22 views

Adjusting corner frequency to constrain maximum disturbance in a high-pass filter

I have a first-order high-pass filter with transfer function: $$G(f)=\dfrac{G_0 jf}{jf + f_c}$$ where $G_0$ is the gain at high frequencies. If I input a sine wave with frequency 1 KHz and I want a ...
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115 views

Biquad Filter Optimization

I would like to try optimize a bit more the biquad filters indended for RIAA/non-RIAA equalization at low samplerates (44.1/48kHz). Best fit I get now is little below ±0.3dB for 44.1kHz and below ±0....
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1answer
51 views

What filter to use in audio analysis filterbank instead of FFT?

Standard bandpass filters can make super precise analysis filterbanks with 1024 to 4096 filters, on reaktor4. I tried in code to used cookbook BandPass and the result was aweful. Does someone know a ...
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1answer
41 views

How many non-zeros samples in convolution

If I have $b[n]$ of length 30 (30 non-zero samples) and $c[n]$ of length 40 (40 non-zero samples). How many non-zero samples will $a[n]=b[n] * c[n]$ have? (Note '$*$' is a convolution). I think ...
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414 views

Can two different impulse responses give the same frequency response?

I have two different impulse response each with different length, is it possible that they have the same frequency response?
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35 views

Real digital filter property

I am a beginner to study about the filter notion and property Being a real digital filter, (here "real filter" I means that its impulse response is real-valued) this formula is established. But I ...
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61 views

s_to_z (Pupalaikis)

Paper: Bilinear Transformation Made Easy - http://documents.mx/documents/easybilinearpdf.html Example of implementation - http://codepad.org/u3tvKn0S I get the same output for Butterworth lp example ...